System and method for interacting with a user via a variable volume and variable tone audio prompt

ABSTRACT

A system plays audio prompts to a user as part of a voice application. If the user begins to provide input in response to the audio prompt before the audio prompt is finished playing, the volume of the audio prompt is reduced, but the audio prompt continues to play. If the user finishes providing his input and the system is able to interpret the input before the audio prompt finishes, the audio prompt will halt and the system will take further action in accordance with the user&#39;s input. If the user finishes providing his input while the audio prompt is still playing, and the system is unable to interpret the input, the volume of the audio prompt will increase back to the original volume level until either the audio prompt finishes playing, or the user again begins to provide input.

This application claims priority to the filing date of U.S. Provisional Application No. 61/157,346, which was filed on Mar. 4, 2009, the contents of which are hereby incorporated by reference. This application is also a continuation-in-part of U.S. application Ser. No. 11/514,116, which was filed on Sep. 1, 2006 now abandoned, which itself claims priority to the filing date of U.S. Provisional Application No. 60/712,808, which was filed on Sep. 1, 2005, the contents of both of which are hereby incorporated by reference.

FIELD OF THE INVENTION

The invention relates to systems and methods that allow a user to easily interact with a voice application. More specifically, the invention is related to systems and methods that perform voice applications with varying volume audio prompts and varying tone audio prompts to help guide a user's interactions with the voice applications.

BACKGROUND OF THE INVENTION

One common example of a system that provides voice services is an Interactive Voice Response (IVR) system. In prior art systems, a user would typically use a telephone to call in to a central computer system which provides voice services via an IVR system. The IVR system deployed on the central computer system would then launch voice services, for instance by playing an audio clip containing a menu of choices to the user via the telephone line connection. The user could then make a selection by speaking a response. The spoken response would be received at the central computer system via the telephone line connection, and the central computer system would interpret the spoken response using speech recognition techniques. Based on the user's response, the IVR system would then continue to perform application logic to take further action. The further action could involve playing another menu of choices to the user over the telephone line, obtaining and playing information to the user, connecting the user to a third party or a live operator, or any of a wide range of other actions.

In interactive voice response (IVR) systems as discussed above, once the caller is connected to a system, the system plays the user an audio prompt. Typically, when the system is playing an audio prompt to a user that asks the user to make a selection, issue a command, or provide input, the user is able to begin speaking his response at any time. In other words, the user need not wait until the entire audio prompt has played to begin providing his spoken input. In the past, if the user begins speaking (to provide his input) before the audio prompt has finished playing, the system could halt the playing of the audio prompt, or continue playing the prompt. Regardless of which approach is taken, the system would immediately begin to try to interpret the user's spoken input. Either approach can be problematic.

In the case where the audio prompt is halted as soon as the user begins speaking, there is a danger that the user was speaking for some purpose other than to provide input to the system. For instance, the user might be speaking to another person in the room. If the user's utterance was not intended as input in response to the audio prompt, then halting the playing of the audio prompt inevitably delays or impedes the process. In this instance, the system would try to interpret the user's utterance, and even if the system can successfully interpret the response, the words would not be useful as a response to the audio prompt. As a result, the system would then have to notify the user that the spoken response it received was not understood, and the system would have to begin playing the audio prompt over again from the beginning.

Problems can also arise if the audio prompt simply continues to play as the user begins to speak. If the user was actually providing spoken input in response to the audio prompt, the fact that the audio prompt continues to play can be confusing and disruptive to the user. The user might assume that the system cannot hear him, because the audio prompt did not halt. As a result, the user might stop partway through providing his spoken input. And because the system does not receive all of the intended user input, the system will have to indicate that the response was not understood, and the system would then play the same audio prompt over again from the beginning. All of which can be very frustrating to the user. In fact, it is likely that the second time the audio prompt begins to play, the user will wait for the prompt to completely play before speaking his response, to avoid the problems experienced the first time around. And this will unnecessarily delay the process of receiving the user's input.

In another instance, the user might speak a response to the audio prompt while the audio prompt continues to play, but the response will be unrecognizable to the system. If the system is unable to interpret the response, the system will assume that the utterance was not intended as a response to the audio prompt, but was likely a statement made to a third party while the audio prompt was playing. Based on this assumption, the system will continue to play the audio prompt to completion, and it will then wait to receive the user's spoken input. The user, thinking that he has already given his response, will also wait for the system to take the next action. After a certain period of time has elapsed, the system will provide an indication that no input has been received, and it will then play the same audio prompt over again from the beginning. All of which can be extremely frustrating to the user.

In addition to the above-discussed drawbacks of existing IVR systems, the ability to provide any type of voice services has been quite limited by the nature of the systems that provide such services. In the known systems that provide voice services using relatively complex speech recognition processing, the voice applications are performed on high end computing devices located at a central location. Voice Application processing requires a high end centralized computer system because these systems are provisioned to support many simultaneous users.

Because complex voice application processing must be provided using a high end computer system at a central location, and because users are almost never co-located with the high end computer system, a user is almost always connected to the central computer system via a telephone call. The call could be made using a typical telephone or cell phone over the PSTN, or the call might be placed via a VoIP-type (Skype, SIP) connection. Regardless, the user must establish a dedicated, persistent voice connection to the central computer system to access the voice services.

In a typical prior art architecture for a centralized voice services platform, the speech recognition functions are performed at a central computer system. A user telephone is used to place a telephone call to the central voice services platform via a telephone network. The telephone network could be a traditional PSTN, or a VoIP based system. Either way, the user would have to establish the telephone call to the central voice service platform via a telephone carrier.

The prior art centralized voice services platforms, which depend on a telephony infrastructure for connection to users, are highly inflexible from a deployment standpoint. The configurations of hardware and software are all concentrated on a small number of high end servers. These configurations are technically complex and hard to monitor, manage, and change as business conditions dictate. Furthermore, the deployment of existing IVR system architectures, and the subsequent provisioning of users and voice applications to them, requires extensive configuration management that is often performed manually. Also, changes in the configuration or deployment of IVR services within extant IVR architectures often require a full or partial suspension of service during any reconfiguration or deployment effort.

Further, cost structures and provisioning algorithms that provision the capabilities of such a centralized voice services platform make it virtually impossible to ensure that a caller can always access the system when the system is under heavy usage. If the system were configured with such a large number of telephone line ports that all potential callers would always be connected to access contrasting types of voice services, with different and overlapping peak utilization hours, the cost of maintaining all the hardware and software elements would be prohibitive. Instead, such centralized voice services platforms are configured with a reasonable number of telephone ports that result in a cost-effective operating structure. The operator of the system must accept that callers may sometimes be refused access. Also, system users must accept that they will not receive an “always on” service.

Prior art centralized voice services platforms also tend to be “operator-centric.” In other words, multiple different service providers provide call-in voice services platforms, but each service provider usually maintains their own separate platform. If the user has called in to a first company's voice services platform, he would be unable to access the voice services of a second company's platform. In order to access the second company's voice services platform, the user must terminate his call to the first company, and then place a new call to the second company's platform. Thus, obtaining access to multiple different IVR systems offered by different companies is not convenient.

In addition to the above-described drawbacks of the current architecture, the shared nature of the servers in a centralized voice services platform limits the ability of the system to provide personalized voice applications to individual users. Similarly, the architecture of prior art IVR systems limit personalization even for groups of users. Because of these factors, the prior art systems have limitations on their ability to dynamically account for individual user preferences or dynamically personalize actual voice applications on the fly. This is so because it becomes very hard for a centralized system to correlate the user with their access devices and environment, to thereby optimize a voice application that is tuned specifically for an individual user. Further, most centralized systems simply lack user-specific data.

With the prior art voice services platforms, it was difficult to develop efficient mechanisms for billing the users. Typically, the telephone carrier employed by the user would bill the user for calls made to the voice services platform. The amount of the charges could be determined in many different ways. For instance, the telephone carrier could simply bill the user a flat rate for each call to the voice services platform. Alternatively, the telephone carrier could bill the user a per-minute charge for being connected to the voice services platform. In still other methods, the voice services platform could calculate user charges and then inform the carrier about how much to bill the user. Regardless of how the charges are calculated, it would still be necessary for the telephony carrier to perform the billing, collect the money, and then pay some amount to the voice service platform.

Prior art voice services platforms also had security issues. In many instances, it was difficult to verify the identity of a caller. If the voice services platform was configured to give the user confidential information, or the ability to transfer or spend money, security becomes an important consideration.

Typically, when a call is received at the voice services platform, the only information the voice services platform has about the call is a caller ID number. Unfortunately, the caller ID number can be falsified. Thus, even that small amount of information could not be used as a reliable means of identifying the caller. For these reasons, callers attempting to access sensitive information or services were usually asked to provide identifying data that could be compared to a database of security information. While this helps, it still does not guarantee that the caller is the intended user, since the identifying data could be provided by anybody.

Some prior art voice services platforms were used to send audio messages to users via their telephones. The central voice services platform would have a pre-recorded audio message that needed to be played to multiple users. The platform would call each of the users, and once connected to a user, would play the audio message. However, when it was necessary to contact large numbers of users, it could take a considerable amount of time to place all the calls. The number of simultaneous calls that can be placed by the centralized voice services platform is obviously limited by the number telephone ports it has. Further, in some instances, the PSTN was incapable of simultaneously connecting calls on all the available line ports connected to the voice services platform. In other words, the operators found that when they were trying to make a large number of outgoing calls on substantially all of their outgoing lines, the PSTN sometimes could not simultaneously connect all of the calls to the called parties. Further, when a voice services platform is delivering audio messages in this fashion, they tie up all the telephone port capacity, which prevents users from calling in to use the service.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 illustrates elements of a system embodying the invention;

FIG. 2 illustrates elements of another system embodying the invention;

FIG. 3 illustrates the volume of an audio prompt as a function of time;

FIG. 4 illustrates the volume of an audio prompt as a function of time under other conditions;

FIG. 5 illustrates the volume of an audio prompt as a function of time under other conditions; and

FIGS. 6A and 6B illustrate how the volume of an audio prompt can be reduced to varying levels depending upon the type of input a user provides in response to the audio prompt.

DETAILED DESCRIPTION OF PREFERRED EMBODIMENTS

The inventors have developed new systems and methods of delivering voice-based services to users which make use of some aspects of the basic architecture illustrated in FIG. 1. A full description of the systems and methods created by the inventors is provided in U.S. patent application Ser. No. 11/514,116, which was filed on Sep. 1, 2006.

The systems and methods created by the inventors are intended to provide users with speech and touch tone enabled Voice Applications for accessing various services and for performing various functions. In this respect, the systems, devices and methods embodying the invention serve some of the same functions as prior art centralized voice services platforms. The systems and methods can also be used to provide the same type of call forwarding discussed above, but at a lower cost, and with greater flexibility. In addition, the systems and methods created by the inventors make it possible to provide users with a whole host of additional call handling and call notification functions that would have been impossible with prior systems.

Unlike the prior art voice services platforms, systems and methods embodying the invention utilize a highly distributed processing architecture to deliver the services. As will be explained below, the underlying architecture and the distributed nature of systems and methods embodying the invention allow the inventive systems to provide the same services as the prior art systems, but with better performance, at a significantly reduced cost, and with far fewer limitations. In addition, systems and methods embodying the invention avoid or solve many of the drawbacks of the prior systems. Further, because of the way systems and methods embodying the invention operate, they can provide new and additional services that could never have been provided by the prior art systems. Systems and methods embodying the invention also allow for much better personalization of delivered services, and they allow existing services to be upgraded, improved, or further personalized much more easily than was possible with the prior art systems.

Systems and methods embodying the invention are intended to deliver or provide Voice Applications (hereinafter, “VAs”) for a user. Before beginning a discussion of systems and methods that embody the invention, we should start by discussing what a VA is, and what a VA can do for a user. Unfortunately, this is somewhat difficult, because VAs can take a wide variety of different forms, and can accomplish a wide variety of different tasks.

A VA provides a user with the ability to use their natural voice, touch tone sequences or other forms of user input, to access and/or control an application, to obtain information, to perform a certain function, or to accomplish other tasks. Although the majority of the following description assumes that a user will interact with a system embodying the invention, at least in part, via speech, other forms of user interaction fall within the scope and spirit of the invention. For instance, developing technologies that allow a user to make selections from visual menus via hand or eye movements could also for the basis of a user interaction protocol. Likewise, developing technologies that are able to sense a user's brainwave patterns could form the basis of a user interaction protocol. Thus, systems and methods embodying the invention are not limited to speech-based user interfaces.

A VA could be specifically developed to utilize the benefits of speech recognition-based input processing. For instance, a VA could be developed to access, play and manipulate voice mail via speech commands. Alternatively, a VA could act as an extension or an enhancement of traditional GUI-like applications to allow the traditional applications to be accessed and/or controlled by speech commands. For instance, a VA could allow the user to call up specific e-mail messages on a display via spoken commands, and the user would then read the e-mail messages on the display.

In some instances, a VA could act like one of the interactive voice response systems that are accessible to users on prior art centralized voice services platforms. A VA could act in exactly the same way as a prior art IVR system to allow a user to obtain information or accomplish various functions using a speech enabled interface. However, because of the advantages of the new architecture, a system embodying the invention can perform voice applications that would have been impossible to perform on prior art centralized voice services platforms. Other VAs could perform a wide variety of other tasks. In most instances, the user would be able to accomplish functions or obtain information by simply speaking voice commands.

With the above general description of a Voice Application (VA) as background, we will now provide an overview of systems and methods embodying the invention. The following overview will make reference to FIG. 1, which depicts a high-level diagram of how a system embodying the invention would be organized.

As shown in FIG. 1, preferred embodiments of the invention would make use of an optional telephone network 230 and a data network 220. The telephone network 230 could be a traditional PSTN, a VoIP system, a peer-to-peer telephone network, a cellular telephone network, or any other network that allows a user to place and receive telephone calls. The data network 220 could be the Internet, or possibly a private or internal local area network or intranet.

In some instances, users would only be physically coupled to a data network, such as the Internet. In this case, the user's on-site equipment could enable them to place VoIP telephone calls via the data network. Such VoIP telephone calls might make use of the PSTN, or the entire call might be handled over the data network. Regardless, in preferred embodiments, the user would be capable of simultaneously maintaining a telephone connection and sending and receiving data.

Systems embodying the invention, as shown in FIG. 1, will be referred to as having a Distributed Voice Application Execution System Architecture (hereinafter, a “DVAESA”). Thus, the term DVAESA refers to a system and method of providing voice application services in a distributed fashion, over a network, to a customer device. Such a system is closely managed by a centralized system to, among other things, ensure optimum performance, availability and usability. In some of the descriptions which follow, there are references to “DVAES-enabled” equipment or local devices/device. This means equipment and/or software which is configured to act as a component of a DVAESA embodying the invention.

A user would utilize an audio interface device to access the DVEASA. In the embodiment shown in FIG. 1, a first user's audio interface 200 comprises a microphone and speaker. A second user audio interface 201 comprises a telephone. The telephone 201 is also connected to the same user local device 210 as the first user audio interface. A third user's audio interface 202 could also comprise a telephone. This telephone 202 could be a regular wired telephone, a wireless telephone or even a cellular telephone. The DVAES-enabled devices may support multiple audio interface devices, and the multiple devices could all be of the same type, or multiple different types of user audio interfaces could all be connected to the same local device.

Each user would also make use of a local DVAES-enabled device that would act to deliver or provide VAs to the user through the user's audio interface. The local DVAES-enabled devices would include a voice browser capable of performing voice applications that have been distributed over the network, some of which may have speech recognition functions. Such voice applications could be pre-delivered to the local DVAES-enabled device, or the voice applications could be fetched in real time. Such voice applications are personalized to the user and optimized for the device. In the embodiment shown in FIG. 1, each of the user local devices 210, 212, 203 are coupled to the respective user audio interfaces, and to the data network.

In some embodiments of the invention, a user audio device and a DVAES-enabled device could be integrated into a single electronic device. For instance, a PDA with cell phone capability could also incorporate all of the hardware and software elements necessary for the device to also act as the DVAES-enabled equipment. Thus, a single user device could function as both the DVAES-enabled equipment that communicates with the network, and as the user audio interface. The user local device 203 shown in FIG. 1 is intended to illustrate this sort of an embodiment.

Also, in FIG. 1, various lines connect each of the individual elements. These lines are only intended to represent a functional connection between the two devices. These lines could represent hard-wired connections, wireless connections, infrared communications, or any other communications medium that allows the devices to interact. In some instances the connections could be continuous, and in others the connection could be intermittent. For instance, an audio interface and a user local device could be located within a user's vehicle. In such a case, the local device within the vehicle might only be connected to the network through a cellular telephone network or through another type of wireless network when such connectivity is required to provide a user with services. In a similar embodiment, the local device in the user's vehicle might only link up to the network when the vehicle is parked at the user's home, or some other location, where a wireless connection can be implemented.

Also, the user audio interface 202 shown in FIG. 1 could be a cell phone that is capable of interacting with the normal cellular telephone network. However, the cellular telephone might also be capable of interacting with the user local device 212 via a wired or wireless connection. Further, the cellular telephone 202 might be configured such that it acts like a regular cellular telephone when the user is away from home (and is not connected to the local device 212). But the cellular telephone might switch to a different operating mode when it is connected to the local device 212 (when the user is at home), such that all incoming calls to that cell phone are initially received and processed by the local device 212. The DVAESA also would include some network-based elements. As shown in FIG. 1, the network-based elements could include a VA rendering agent 240, a network storage device 242 and a system manager 244. Each of these network-based elements would be connected to the data network.

Also, although they would not technically be considered a part of the DVAESA, there might also be some third party service providers 250, 252 which are also connected to the data network, and/or to the telephone network. As explained below, the VAs may enable the users to interact with such third party service providers via the data and telephone networks.

When a DVAESA as shown in FIG. 1 is configured, VAs would be “rendered” by the VA rendering agent 240, the output of the rendering process would be rendered VAs. These rendered VAs may be stored on the Network Storage Device 242, or be distributed or delivered to a DVAES-enabled Device. “Rendering” refers to a process in which a generic VA is personalized for a particular user and/or one or more particular DVAES-Devices to generate Rendered VAs. The system manager 244 could instruct the VA rendering agent 240 to render a VA for a particular user, or such rendering request could originate from the DVAES-enabled Device. The DVAESA network data storage element 242 could be used to store generic VA, rendered VAs, or a wide variety of other data and resources (e.g. audio files, grammars etc).

As mentioned above, the VA rendering agent would personalize a generic VA during the rendering process. This could take into account personal traits of the individual user, information about the configuration of the local device(s), or a wide variety of other things, as will be explained in more detail below. The information used to personalize a VA during the rendering process could be provided to the VA rendering agent at the time it is instructed to render the VA, or the VA rendering agent could access the information from various data storage locations available via the data network.

The user's local devices would typically be inexpensive computing devices that are capable of running a voice browser and performing speech recognition capable rendered VAs. Such devices are often referred to as embedded multimedia terminal adaptors (EMTAs) and optical embedded multimedia terminal adaptors (OEMTAs). In many instances, the local device would be physically present at the user's location, such as a home or office. In other instances, however, the local device could be a virtual device that is capable of interacting with one or more user audio interfaces. As mentioned above, the local devices may also store rendered VAs, and then act to perform the rendered VAs to the user's audio interface. The user local device could be a customer premise device that is also used for some other function. For instance, the local device could be a cable modem or set-top box that is also used to connect a television to a cable network, however, the device would also be configured to perform VAs for the user via the user's audio interface.

In one simple embodiment of the invention, a local embedded device 212 would be linked to a user's telephone 202. The local device 212 would also be linked to the Internet 220 via a medium to high speed connection, and possibly to the telephone network 230. The user could speak commands into the telephone 202, and those spoken commands would be processed by the local device 212 to determine what the user is requesting.

The processing and interpretation of a user's spoken commands could be entirely accomplished on the local device 212. In other embodiments, the local device might need to consult a speech recognition engine on a remote device, via the data network, to properly interpret a portion of a spoken command that cannot be understood or interpreted by the local device. In still other embodiments, the user's spoken commands could be entirely processed and interpreted by a remote speech recognition engine. For instance, a recording of the user's spoken commands could be relayed to a remote speech recognition engine, and the speech recognition engine would then process the spoken commands and send data back the local device indicating what the user is commanding. Even this process could be accomplished in real time such that the user is unaware that the interpretation of his spoken commands is being accomplished on a remote device.

Because of the greater sophistication that is possible with a system embodying the invention, if the local device does not understand something, it can often ask another question of the user to clarify the situation. In addition, the local device can offer greatly expanded vocabulary and speech processing by enlisting the assistance of network agents. For all these reasons, a consumer electronic device that is coupled into the DVAES architecture can provide a much more sophisticated voice application than prior art devices which were not connected to a network.

Once the spoken command has been interpreted, in some instances, the local device 212 may be able to satisfy the user's request. In other instances, the local device 212 might need to request information from a VA Rendering Agent 240 to satisfy the user's request. If that is the case, the local device 212 would send a query over the data network 220 to the VA Rendering Agent 240 for some type of content. The requested content would be returned to the local device 212, and the local device 212 would then provide the content to the user via the user's telephone 202. In other instances, the local device may be able to query other network-connected elements which are not a part of the DVAES Architecture, and those other elements would return the requested data to the local device so that the data could be delivered to the user via the audio interface.

Depending on the VA being performed, the functions that are performed in response to a user request may not involve playing audio information to the user via the user's audio interface. For instance, the local device could be performing a VA relating to accessing e-mail. In this instance, a user's spoken request could cause the local device to act in a manner that ultimately results in the user's e-mail messages being shown on a display screen. In this instance, although the user makes use of a speech-based interface to obtain information and/or perform a certain function, the ultimate result is not the playback of audio, but rather display of an e-mail message.

The end result of a user request could take many other forms, such as the local device causing a certain action to be taken. For instance, the user might speak a request that causes the user's home air conditioning system to be turned on. The list of possible actions that could be enabled by the local device is virtually endless. But the point is that the local device is able to provide a speech-enabled interface to the user, via the audio interface, to allow the user to accomplish a task.

In another simple embodiment, the user might pick up his telephone 202 and speak a request to be connected to another person's telephone. A voice application performed on the local device would interpret the user's spoken request. This could be done on the local device, or the voice application could utilize remote assets to accomplish the speech recognition. Some or all of the speech recognition could occur on the remote assets. The voice application would then take steps to place a telephone call to the person identified by the user. This might involve connecting the user via the telephone network 230, or connecting the user to the requested party via a VoIP call placed over the data network 220.

It is also worth noting that when a user is connected to the DVAES architecture, the VAs provided by the system can completely replace the dial tone that people have come to associate with their telephones. The moment that a user picks up his telephone, he will be launched directly into a voice application that is provided by the system. In the past, this may have been technically possible, but it was always accomplished by making use of the traditional phone system. For instance, one of the prior art centralized voice services platforms would have been capable of ensuring that the moment a user lifts his telephone, that user was immediately connected to a central voice services platform that would guide the remainder of the user's experience. But this was always accomplished by establishing an immediate voice channel between the user's telephone and the central voice services platform. And to accomplish that, it was necessary to involve the telephone carrier that would link the user's telephone to the voice services platform. In contrast, with the DVAES architecture, one no longer needs to make any use of the telephone carriers to provide this sort of a service. And, as noted above, the user can still be easily connected to the regular telephone network if he needs to place a call.

In the same vein, in the past, whenever a user wanted to have a third party service answer his telephone calls, as in traditional voice mail systems, it was necessary to involve the carrier in routing such calls to a third party service. Now, when a call is made to the user's telephone, the DVAES architecture makes it possible to answer the call, and take voice mail recordings, without any further involvement of the carrier. Here again, the DVAES architecture makes it possible to eliminate the services of the telephone carrier.

In both the examples outlined above, the involvement of the carrier necessarily increased the cost of providing the voice services. Because the carrier can be eliminated, the same sorts of voice services can be provided to a user for a significantly reduced cost. And, as explained below, the services can be delivered with greater performance and with new and better features.

In some embodiments, rendered Voice Application processing is performed on the local device and the associated the voice recognition functions may also be performed on the local device. For this reason, there is no need to establish a dedicated duplex audio link with a remote high end computer. Also, even in those instances where a portion of the voice application processing is performed by a remote device, and/or where processing and interpretation of spoken commands is processed by a remote device, the communications necessary to accomplish these actions can be made via data packets that traverse a data network. Thus, here again, there is no need to establish a dedicated duplex audio link with a remote high end computer to provide the requested services.

Also, because the local embedded device is coupled to a data network such as the Internet, it can rapidly obtain Rendered Voice Applications and associated data from various remote sources in order to satisfy user requests. For these reasons, the simple embedded local device allows one to provide the user with speech recognition enabled Voice Applications without the need to create and maintain a high end speech service platform with multiple telephone line access equipment.

As noted above, the local device could also use the network to obtain access to various other physical elements to effect certain physical actions, such as with the home air conditioner example given above. In this context, the other physical elements could be connected to the network, or the local device could have a local connection to physical elements that are also located on the user's premises. For instance, the local device could have a hard-wired or wireless connection to many different elements in a user's home or office that allow the local device to control operations of the physical elements. In other embodiments, the piece of physical equipment could act as the local device itself.

One obvious advantage of a DVAESA over prior art voice service platforms is that a DVAESA embodying the invention can provide VAs to users without any involvement of a PSTN, VoIP, Peer-Peer carrier. The instant the user picks up his telephone handset, he will be interacting with the DVAESA, not the telephone system. A large number of VAs could be accomplished without ever involving a telephone carrier as the Voice Application is delivered and provided on the local device. Because the user can directly access the DVAESA without making a telephone call, the operator of the DVAESA will not need to pay a telephone carrier in order to provide the service to users.

As noted above, if the user wishes to place a telephone call, this can be easily accomplished. But there is no need to use a telephone carrier as an intermediary between the user and the DVAESA. This has multiple positive benefits.

Also, for a multitude of different reasons, a DVAESA will be less expensive to deploy and operate than the prior art central voice services platforms. To begin with, because the DVAESA can provide services to users without a telephone link, the DVEASA operator no longer need to purchase and maintain multiple telephone line ports into the system.

Also, the types of equipment used by the DVAESA are inherently less expensive to deploy and manage than the equipment used in a central voice services platform. A DVAESA embodying the invention uses relatively inexpensive network appliances that can be located anywhere, and that can be deliberately distributed over a wide area to enhance reliability of the system. In contrast, a central voice services platform requires expensive and specialized telecom equipment like telecom switches and IVR servers. The central voice services platforms also require more intensive management and provisioning than a DVAESA, and this management must be provided by highly skilled personnel as most of the equipment used is highly proprietary in nature. In contrast, the DVAESA is largely managed by an automated management system.

A prior art central voice services platform is only able to simultaneously service a limited number of users. As noted above, in the prior art central voice services platforms, a dedicated voice link, via a telephone call, is maintained for each connected user. Once all lines are connected to users, no additional users are able to access the system. Hence the maximum number of simultaneous users that can be supported at any given time is equal to the lesser of the number of access lines or the number of associated telephony/IVR ports an operator maintains.

In contrast, a DVAESA embodying the invention has a very high limit on the number of users that can be simultaneously serviced. In a DVAESA embodying the invention, the moment a customer picks up his telephone he will be connected to the system. Thus, a DVAESA embodying the invention is “always on.” Also, much of the interactions between the user and the system are handled directly by the local device on the customer premises. If the local device cannot immediately service a user request, and additional information is needed, the local device may make a synchronous or asynchronous request over the Internet. Typically, the information will be quite rapidly returned and played to the user. Thus, even if there is a small delay, the user is nevertheless still connected the voice services system.

With the DVAESA model, the same number of server assets can handle data requests from a much larger number of users as compared to the prior art central voice services platform. This is also another reason why a DVAESA is less expensive to deploy and maintain than a prior art central voice services platform.

In addition to being easier and less expensive to deploy and maintain, a DVAESA embodying the invention can also scale up much more quickly and at a lower cost as new users are added to the system. To begin with, because the DVAESA does not require dedicated telephone lines to operate, there is no cost associated with adding additional telephone ports to the system to accommodate additional users. Likewise, as new users are added, there are no new additional telecommunications expenses for more connect time or access. In addition, for the reasons noted above, the equipment used by the system is far less expensive than the equipment used in a central voice services platform to service the same number of users. Thus, adding any new equipment and users is less expensive for a DVAESA. Moreover, because it requires less equipment to service the same number of users in a DVAESA, there is much less equipment to purchase and maintain for each additional 1000 users.

A DVAESA embodying the invention is inherently more reliable than a prior art central voice services platform. Because the assets of a prior art system are typically located in a few physical locations, and are tied to physical phone lines, power outages and other physical problems are more likely to prevent users from being able to use the system. In contrast, a DVAESA can have its equipment distributed over a much wider area to reduce these problems. The points of a failure of a DVAESA can be highly localized and it is very cost effective to replicate DVAESA equipment.

Moreover, the underlying nature of the DVAESA makes it easy to connect multiple redundant servers to the network, so than in the event one or more assets fail, redundant assets can step in to take over the functions of the failed equipment. This was difficult to do in prior art central voice services platforms, and even when it was possible to provide redundant capabilities, the cost of providing the redundant equipment was much higher than with a DVAESA.

In addition, a prior art central voice service platform needs a telephone carrier to provide access to the users. If the telephone carrier has a service outage, the prior art system cannot function. In contrast, a DVAESA does not have any reliance on a telephone carrier.

The only network required to provide the DVAESA is the data network like the Internet. The user in most cases will not experience an interruption to access to the voice services of a DVAESA, even if there is an outage that disables the local device's access to the Internet. The local device could potentially perform some of the applications without connecting to the network. This indicates that for some Voice Applications in the DVAESA, it may be sufficient for the local device to have intermittent access to the Internet.

The architecture of a DVAESA makes it inherently able to deliver certain types of VAs with vastly improved performance. To use one concrete example, as noted above, when a central voice services application is attempting to deliver the same audio message to large number of users, the central voice services application must place a telephone call to each user, using a dedicated phone line, and deliver the message. Because the central voice services platform only has a limited number of outgoing lines, it can take a significant amount of time to place all those calls.

In contrast, in a DVAESA embodying the invention, it is not necessary to place any telephone calls to deliver the audio message to users. Instead, a server which is part of the system can push instructions to play the audio message, and the message itself (the message could be stored in advance of when the event to deliver the message occurs), to each of the local devices, and the local devices can then play the messages for each individual user. In variations on this theme, the server might only send the instruction to play the message, along with a reference to where a copy of the audio message is stored. Each local device could then download a copy of the message from the indicated location and play it for the user. Regardless, it would be possible for the DVAESA architecture to deliver the audio message to all the users in a small fraction of the time that it would take the prior art central voice services platform to accomplish the job.

Moreover, as also explained above, while the prior art central voice services platform is making calls to deliver audio messages to a plurality of users, it is tying up it's phone lines, and thus it's capacity to allow users to call in for services. In contrast, when a DVAESA is delivering audio messages to a plurality of users, the users are still able to access their voice services for other purposes.

A DVAESA embodying the invention also makes it possible to deliver many new voice applications and services that could never have been provided by the prior art central voice services platform. In most cases, it is the underlying differences in the architecture of a DVAESA embodying the invention, as compared to the prior art voice services platforms, which make these new services possible.

For example, a user could configure a voice application to run constantly in the background on a local device, and then take a certain action upon the occurrence of a specified event. So, for instance, the user could set up a voice application to break into an existing telephone conversation to notify him if a particular stock's trading price crosses a threshold. In this scenario, the voice application would periodically check the stock price. If the threshold is crossed, the voice application could cause any existing telephone call that the user is on to be temporarily suspended, and the voice application would then play the notification. The voice application could then return the caller to his call. This sort of a voice application would also be very complicated to provide under the prior art central voice services platform.

The graceful integration of advertising messages is another example of how a DVAESA embodying the invention can provide services that were impossible to provide with prior art central voice service platforms. As an example, if the user lifted the telephone and spoke a command that asked for options about ordering a pizza, the system could respond with a prompt that said, “to be connected to Pizza Shop A, say one; to be connected to Pizza Shop B, say two. By the way, Pizza Shop A is having a two for one special today.” Thus, the advertising message could be gracefully incorporated into the played response. Also, the advertising message would be highly context relevant, which would make it more interesting to advertisers. Thus, advertising revenue could be collected by the operator of the DVAESA system.

A DVAESA embodying the invention could also be used to rapidly collect data from a very large number of users in ways that would have been impossible with prior art central voice services platforms. In this example, assume that a television program is currently airing, and during the program, viewers are invited to vote on a particular issue. In prior art systems, the users would typically place a telephone call to a central voice services platform and make a voice vote. However, as noted earlier, prior art voice services platforms are only able to talk to a limited number of callers at the same time because the callers must be connected by dedicated phone lines.

In a DVAESA embodying the invention, the user might be able to pick up the phone and say, “I want to vote on issue X.” The system would already know that viewers of a television program had been invited to place a vote, so the system could immediately take the user's voice vote. The system could also tabulate the votes from all users making similar voice votes, and then provide the voting results to the television show producers in real time. Because so little actual information is being exchanged, and the exchanges are made over the Internet, thousands, and perhaps even millions of votes could be received and tabulated in a very short period of time. This would have been impossible with prior art central voice services platforms. Furthermore, a DVAES can distribute a fully featured voice application that not only plays the message, but further solicits feedback from the user, optionally tailors the interaction with the user, and may record any user feedback or responses. Furthermore, if the producers of the television show were willing to pay a fee to the operator of the DVAESA, the system could be configured such that as soon as viewers are invited to cast a vote, and for the duration of the voting period, anytime that a user of the DVAESA picks up his telephone to access the system, the system would first respond with the question, “would you like to vote on issue X?” This would be yet another way to derive advertising or promotional revenue from the DVAESA.

There are countless other ways to exploit the architecture of a DVAESA embodying the invention to accomplish tasks and to perform VAs that would have been impossible using the prior art central voice services platforms. The above examples are merely illustrative.

A DVAESA embodying the invention also allows for much greater personalization of the voice applications themselves than was possible with prior art central voice services platforms. In addition, the architecture allows the users themselves to control many aspects of this personalization.

To begin with, as explained above, in a DVAESA a VA Rendering Agent is responsible for customizing voice applications, and then delivering the customized voice applications to the local devices at the customer sites. Thus, the basic architecture assumes that each user will receive and run personalized versions of voice applications. This difference alone makes it much, much easier to provide users with personalized voice applications than prior art central voice services platforms.

The VA Rendering Agent could personalize a voice application to take into account many different things. For instance, the VA Rendering Agent could access a database of user personal information to ensure that a VA takes into account things like the user's name, his sex, age, home city, language and a variety of other personal information. The VA Rendering Agent could also access information about the capabilities of the local device at the customer's location that will be providing the VA, and possibly also the type of audio interface that the user has connected to the local device. The VA Rendering Agent could then ensure that the customized version of the VA that is provided to the user's local device is able to seamlessly and efficiently run on the local hardware and software. The VA Rendering Agent could also take into account user preferences that the user himself has specified. For instance, the VA could be customized to play audio prompts with a certain type of voice specified by the user.

Another important way that VAs could be personalized is by having the DVAESA track how the user is interacting with the system. For Example if the user has a certain type of accent or has a certain pattern of use or has a certain type of background noise, the VA Rendering Agent could take these factors into account on an on going basis to ensure that the customized VAs that are sent to the user are tuned to the user. The system might also note that whenever a three choice menu is played to the user, the user always makes the third selection. In that case, the VA Rendering Agent might be directed to re-render the VA so that the VA presents the third option first, instead of last.

There are any number of other ways that VA's could be customized or personalized to take into account aspects of individual users. And these customizations are easily and automatically accomplished by configuring the VA Rendering Agents to automatically incorporate these personalizations when delivering VAs for users. Because the DVAESA is configured so that each individual user may have his own versions of VAs, preferably stored on his local devices cache, this personalization is not difficult to accomplish. Such personalizations are complimented by the continuous analytics process that is being performed on DVAESA data. This data is collected during the on going functioning of the system and is provided by all DVAESA components. After collection, the data is analyzed, and the results of the analysis are used to continuously tune and improve the functioning of the system on an individual user-by-user basis.

A DVAESA also allows for better, more direct billing for delivery or usage of services. Because there is no telephone company acting as an intermediary, the operator of a DVAESA can directly bill users for use of the system. Also, the way the system is configured, the user can select individual services, which are then provided to him by rendering a VA and loading it on the user's local equipment. Thus, the user can tailor his services to his liking, and the operator of the DVAESA has an easy time tracking what services the user has. For all these reasons, it is much easier to bill the user for use of the services.

Another benefit that flows from the DVAESA model is the ability of a user to access services provided from two different DVAESA operators on a single piece of local equipment. As will be explained in more detail below, a first DVAESA operator could load a first set of VAs onto the user's local equipment, and a second DVAESA operator could load a second set of VAs onto the same piece of operator equipment. For instance, the first DVAESA operator could be one that provides the user with services related to his business, and the second DVAESA operator could be one that provides the user with services relating to the user's personal life. There is no inherent conflict in both having two different sets of VAs loaded onto the local device. And each DVAESA operator can thereafter maintain and update their respective VAs. Likewise, the user can cause both sets of VAs to be loaded on a first device at his office, and a second device at his home. This allows the user to easily and immediately access services from either operator, regardless of his present location. This sort of flexibility would also have been completely impossible in prior art central voice services platforms.

A DVAESA can also provide enhanced security measures compared to prior art central voice services platforms. For instance, because the DVAESA is interacting with the user via spoken commands, it would be possible to verify the identity of a user via a voice print comparison.

In addition, the individual local devices can be identified with unique ID numbers, and credentials verifying the identity and permissions of users and devices can all be created and stored in various locations on the system. By using these unique identification numbers and certification files, one can ensure that only authorized users can access sensitive information or perform sensitive functions.

Having now provided a broad overview of the how a system embodying the invention would operate, and the inherent advantages of a DVAESA system as compared to prior art systems, we will now turn to a slightly more specific description of the main elements of a DVAESA embodying the invention, with reference to FIG. 2. In doing so, we will introduce some new definitions and terminology which will be used throughout the remainder of the detailed description.

A DVAESA would be configured to deploy and utilize one or more Voice Application Agents (hereinafter “VAAs”) which themselves enable the delivery or performance of a VA through a local device that would typically be located in a user's home or office. In some instances, a VAA may be wholly resident on a single local device. In other instances, the functions of a VAA may be split between multiple portions of the overall system. Likewise, a single local device may only host one VAA. Alternatively, a single local device may host multiple VAAs. These variations, and the flexibility they provide, will be discussed in more detail below. The important concept is that a VAA is the agent that is responsible for delivering or performing a VA for the user.

The network 2130 shown in FIG. 2 could be the Internet. However, in some instances, the network 2130 could be a public or private local network, a WAN, or a Local Area Network. In most instances, however, the network 2130 will be the Internet. Also, the network 2130 could also comprise portions of the PSTN, existing cellular telephone networks, cable television networks, satellite networks, or any other system that allows data to be communicated between connected assets.

The devices 2110 and 2120 appearing in FIG. 2 would be the local embedded devices that are typically located at a user's home or office. As shown in FIG. 2, in some instances, a local device 2110 could simply be connected to the user's existing telephone. In other instances, the local device could be coupled to a speaker 2007 and microphone 2009 so that the local device can play audio to the user, and receive spoken commands from the user. In still other embodiments, the local device may be a standalone telephone, or be included as part of a cellular telephone, a computing device with wireless access, a PDA that incorporates a cellular telephone, or some other type of mobile device that has access to a data network.

A system embodying the invention also includes components that deliver voice applications, data and other forms of content to the local devices. These components could include one or more Voice Application Services Systems (hereinafter VASSs). In the system depicted in FIG. 2, there are two VASSs 2140 and 2150. A system embodying the invention could have only a single VASS, or could have multiple VASSs.

One of the primary functions of a VASS is to render VAs and to then provide VA components to VAAs. In preferred embodiments, a VASS would provide customized VAs components to VAAs, upon demand, so that the VAAs can perform the customized VAs components for the user. The VASSs could personalize generic VAs based on known individual user characteristics, characteristics of the environment in which the VA components will be performed, information about how a user has previously interacted with the system, and a wide variety factors. The distribution of the personalized VA components to the VAAs could also be accomplished in multiple different ways.

A system embodying the invention may also include one or more Content Distribution Services (hereinafter a “CDSs”). This is an optional component that basically serves as a data storage and content distribution facility. If a system embodying the invention includes one or more CDSs, the CDSs would typically provide network-based caching of content, such as VA components, configurations, DVAESA components, and other shared or frequently used content. The CDSs would be deployed throughout the network to help reduce network traffic latency, which becomes particularly noticeable in any speech interaction system.

The DVAESA components could broadly be identified as a Distributed Voice Application Execution System (hereinafter, a “DVAES”), and a Distributed Voice Application Management System (hereinafter, a “DVAMS”) A DVAES comprises at least a VASS, one or more VAAs, and the underlying hardware and software platforms.

The system shown in FIG. 2 includes a DVAMS. The DVAMS handles a wide variety of management functions which include registering users, specific items of hardware and other DVAES components, directing the rendering, caching, distribution and updating of VAs components, organizing and optimizing the performance of system assets, and multiple other functions. The DVAMS may also include an interface that allows an individual user to customize how the system will interact with him, and what products and services the user wishes to use. The DVAMS would also provide an interface that allows system operators to manually control various aspects of the system.

As explained above, some voice applications performed by a system embodying the invention may seem to the user like a typical interactive voice response systems in that a voice application would typically play an audio prompt to the user, and the voice application would then receive and interpret the user's response. And as such, VAs will be subject many of the problems with IVRs outlined above in the Background section. Specifically, when a user wishes to begin providing spoken input before an audio prompt is finished playing, the user and/or the system may become confused about when the user's utterances are in response to the audio prompt and whether the system is actually listening to the user when the user begins to speak. The inventors have developed new techniques and methods of performing voice application that at least partially mitigate these problems.

In the new systems and methods developed by the inventors, the audio prompts that are part of voice applications are played at varying volumes and/or with varying background tones. The differences in the volume of a prompt or in the tone that a user hears will help to guide the user in interacting with a voice application.

In some embodiments, an audio prompt would play at normal volume until a user begins to provide spoken input. As soon as the system detects that the user is providing spoken input, the system could halt the audio prompt. However, in other embodiments, if the audio prompt has not finished playing when the system detects that the user has begun to speak, the audio prompt will continue to play. But as soon as the system detects that the user has begun to speak, the volume of the audio prompt will be reduced. The reduction in the volume of the audio prompt will provide an indication to the user that the system recognizes that the user has begun to provide spoken input. Thus, the user will not be confused into thinking that the system has failed to recognize the beginning of the user's input.

In some embodiments, a voice application would begin to play an audio prompt to a user, and as soon as the system detects that the user has begun to speak, the original audio prompt would halt, and a new audio file would be played. The new audio file could continue to play until the user finishes providing his spoken input, and when the system recognizes that the user has finished providing input, the system could halt the new audio file and begin to play a second different audio file.

In other embodiments, a voice application might begin to play an audio prompt, and when the system detects that the user has begun to provide spoken input, the system could continue to play the audio prompt, but at a reduced volume. And when the system detects that the user has finished providing input, the system could halt the audio prompt and begin to play an audio file.

In still other embodiments, the system might be playing either the original audio prompt (at the original or a reduced volume) or an audio file while the user provides spoken input. When the system detects that the user has finished providing spoken input, the system could change the volume level of whatever is playing or the system could halt what was playing while the user provided input and the system could begin to play a different audio file. Once the system has finished processing the user's input, the voice application could take an action based on the input, or the system could again change the volume level of what was playing, or switch to a different audio file.

The common concept through most of these embodiments is that upon the occurrence of an event, either the volume of what is being played changes, or the system switches from playing one thing to playing something else. The something else could even be silence.

The events which could trigger a change in what is being played include the moment that the system detects that the user has begun to provide input, the point in time that the system determines that the user has finished proving input, and the point in time when the system has finished processing a user's input, and the point in time when an audio prompt finishes playing. Any of these events could trigger the voice application to change the volume of what is playing or to change the audio that is playing, or to halt all audio.

Of course, in some voice applications, the input provided by the user may not be spoken input. For instance, the user might type in a response on a keyboard connected to a local device, or the user might push buttons on a telephone keypad that is connected to a local device. In other instances, a user might operate a pointing device to make a selection of an item on a display screen. Various other forms of input might also be provided by a user in response to an audio prompt from a voice application. Regardless of the type of input provided by the user, the system would reduce the volume of an audio prompt that is playing as soon as the system recognizes that the user has begun to provide input.

FIG. 3 illustrates the volume of an audio prompt that is played by a voice application as a function of time. When the audio prompt begins to play at time T₀, the audio prompt is played at high volume V₁. In this instance, the user begins to provide input in response to the audio prompt at time T₁. Thus, at time T₁ the volume of the audio prompt is reduced to volume V₂. The audio prompt will continue to play at volume V₂ until it ends at time T₂.

In the example illustrated in FIG. 3, the user might be continuing to provide his input until after the audio prompt finishes at time T₂. Or, the user may have finished providing his input, but the system has not been able to finish processing the input until after the audio prompt is fully played. As a result, the audio prompt will play to completion at the lower volume.

As soon as the user begins to speak, the system will try to interpret the user's input. If the system is successful in processing the user's input, the system will stop playing the audio prompt (at the lowered volume), and it will take the action indicated by the user's response. FIG. 4 illustrates the volume of the audio prompt as a function of time for this alternate set of events. In the example illustrated in FIG. 4, the same audio prompt is being played as was played in the example illustrated in FIG. 3. However, before the audio prompt can finish playing the audio input provided by the user has been successfully interpreted. Thus, at time T₃ the audio prompt is halted, and the system will take further action in accordance with the user's input.

In some embodiments, if a user provides a complete piece of input while an audio prompt of a voice application is playing, and the system tries but fails to interpret the user's input while the audio prompt is still playing (at reduced volume), the system will increase the volume of the audio prompt back up to the normal level for the remainder of the audio prompt. If the user hears the volume increase back up to the normal level, this will provide an indication to the user that the system was unable to interpret the user's input. Thus, the user will know to re-provide his spoken response.

FIG. 5 shows the volume of an audio prompt as a function to time for this type of situation. Here again, the audio prompt is the same audio prompt that was played in the example illustrated in FIG. 3. As explained, when the audio prompt begins to play at time T₀, the audio prompt is played at high volume V₁. The user begins to provide input in response to the audio prompt at time T₁. Thus, at time T₁ the volume of the audio prompt is reduced to volume V₂. The user finishes his input at time T₄. The system then attempts to interpret the user's input. At time T₅, the system decides that it cannot interpret the user's input. Thus, at time T₅ the system increases the volume of the audio prompt back up to volume V₁. And the audio prompt continues to play at the high volume V₂ until the audio prompt ends at time T₂. The fact that the volume of the audio prompt increased back to the higher volume V₂ signals the user that the input he provided was not interpreted. Thus, the user should know to re-provide his input.

In some instance, the user may have provided a spoken input in response to the audio prompt, and the voice application was simply unable to interpret the response. In other instances, the spoken input provided by the may not have made sense to the system given what the system was asking for, so the system decided to not take any action. Thus could occur if the user speaks while the audio prompt is playing, but the spoken utterance was not made in response to the audio prompt, but rather to a third party located near the user. Regardless, once the volume increases back up to the higher level, the user will know that the system is still waiting for a response to the audio prompt.

The user still does not have to wait for the audio prompt to finish playing. And if the user begins to speak a second time before the audio prompt finishes playing, the volume of the audio prompt will again be lowered while the system tries to interpret the new spoken input.

When a voice application operates as explained above, with varying volume audio prompts, the user will always have an indication of what the system is attempting to do in receiving and processing user input. This will reduce the number of instances where the system is waiting to receive user input, while at the same time, the user is waiting for the system to take an action. This should also reduce the number of instances where an audio prompt must be replayed because a user response was not properly received or interpreted.

Again, it is important to note that user input in response to an audio prompt could take many forms. Providing spoken input is only one form of input. And anytime that a user begins to provide input before an audio prompt is finished playing, the volume of the audio prompt could be lowered as discussed above.

In other embodiments, the volume of an audio prompt might only be lowered when the user begins to provide spoken input. If the user instead begins to provide input through some other means, such as with a keyboard or keypad, or with a pointing device, the volume of the audio prompt might remain at the original high level until the system correctly interprets the input and halts the audio prompt so that it can take action in accordance with the input, or until the audio prompt finishes playing. Thus, a reduction in the volume of an audio prompt might only occur when a user begins to provide spoken input.

Moreover, the ability to cause audio prompts to reduce in volume might be a setting that is controllable by a user. For instance, the user might be able to specify that audio prompts should reduce in volume any time that the user begins to provide spoken input which an audio prompt is playing but not when the user begins to provide input through alternate means. The user might also be able to specify that the volume of an audio prompt be reduced anytime that a user begins to provide input, regardless of the form of the input. Thus, the user could control how this feature is implemented.

Further, the user could specify the actual volume levels that are used in implementing these features. For instance, the user could specify that the volume of an audio prompt should be reduced to 33% of full volume when the user begins to provide spoken input, but that the volume should be reduced to 66% of full volume when the user begins to provide input through other means.

FIGS. 6A and 6B illustrate the concept of different degrees of volume reduction depending on the type of input being provided by the user. In both of these examples, the audio prompt is the same audio prompt as was played in the example illustrated in FIG. 3. In FIG. 6A, the user begins to provide spoken input at time T1. Thus, the volume of the audio prompt is reduced to volume level V3, which is 33% of full volume. In the example illustrated in FIG. 6B, the user begins to provide types input at time T1. Thus, the volume of the audio prompt is reduced to volume level V4, which is 66% of full volume.

Of course, the user could specify that any type or degree of volume reduction should occur whenever the user provides a specific type of input while an audio prompt is still playing. The examples given above are only illustrative and are not intended to be limiting.

In other embodiments of the invention, a voice application could play one or more tones, in addition to an audio prompt, to help guide the user through the voice application. For instance, a voice application would play an audio prompt in the form of a short question, such as “state your name.” As soon as the question finishes playing the system could begin to play a background tone or some background music. The background tone or music would provide an indication to the user that the system is waiting to receive input from the user.

If a background tone or background music is played after a question is posed to the user, the instant that the user begins to provide spoken input, the volume of the background tone or music could be reduced, in a manner similar to that volume reduction discussed above. The reduction in volume would provide an indication to the user that the system recognizes that the user is providing input in response to the question.

In some embodiments, if a user begins to provide spoken input, and then pauses, the background tone could increase back up to the original volume level to provide an indication that the system is waiting to receive additional input from the user. When the user begins to speak again, the background tone could again reduce in volume.

In alternate embodiments, one a user has paused for a sufficient amount of time after providing spoken input, the background tone could increase in volume again to indicate that the system assumes that the user has finished providing input, and the system is processing the input.

In some embodiments, where a repeating tone or background music is played to the user, the tempo of the repeating tone or music could also be varied to provide an indication to the user that certain things are happening. For instance, if the tempo changes from a first speed to a second lower speed, this might provide an indication that the system is processing input provided by the user. A change in tempo could also be used to signal other things.

Various combinations of changes in volume, a switch from one tone to another, changes in the tempo of an audio recording, and various other aspects can be used to provide cues or signals to the user about what the system is doing, and about whether the system is waiting for the user to take further action. Thus, one or more of these aspects can be used alone or in combinations to guide the user through an interaction with a voice application.

Any reference in this specification to “one embodiment,” “an embodiment,” “example embodiment,” etc., means that a particular feature, structure, or characteristic described in connection with the embodiment is included in at least one embodiment of the invention. The appearances of such phrases in various places in the specification are not necessarily all referring to the same embodiment. Further, when a particular feature, structure, or characteristic is described in connection with any embodiment, it is submitted that it is within the purview of one skilled in the art to effect such feature, structure, or characteristic in connection with other ones of the embodiments.

Although the invention has been described with reference to a number of illustrative embodiments thereof, it should be understood that numerous other modifications and embodiments can be devised by those skilled in the art that will fall within the spirit and scope of the principles of this disclosure. More particularly, reasonable variations and modifications are possible in the component parts and/or arrangements of the subject combination within the scope of the foregoing disclosure, the drawings and the appended claims without departing from the spirit of the invention. In addition to variations and modifications in the component parts and/or arrangements, alternative uses will also be apparent to those skilled in the art. 

What is claimed is:
 1. A method of playing audio to a user when a voice application is being performed for the user, comprising: beginning to play the audio in a first fashion; detecting when the user begins to provide input; and continuing to play the audio without interruption, but playing the audio in a second fashion that is different from the first fashion when it is detected that the user has begun providing input and before the input is interpreted or checked for validity.
 2. The method of claim 1, wherein detecting when the user begins to provide input comprises detecting when a user begins providing spoken input.
 3. The method of claim 2, further comprising attempting to interpret the user's spoken input.
 4. The method of claim 3, wherein if the user's spoken input is successfully interpreted, the method further comprises stopping the playing of the audio.
 5. The method of claim 3, wherein if the user's spoken input cannot be interpreted, and the audio is still playing, the method further comprises again playing the audio prompt in the first fashion once the determination has been made that the user's spoken input cannot be interpreted.
 6. The method of claim 1, further comprising: detecting when a user begins to provide non-spoken input; and playing the audio in a third fashion that is different from the first and second fashions when it is detected that the user has begun providing non-spoken input.
 7. The method of claim 6, wherein the non-spoken input is provided via a keyboard or a pointing device.
 8. The method of claim 1, wherein detecting when the user begins to provide input comprises detecting when the user begins to provide non-spoken input.
 9. The method of claim 1, wherein playing the audio in the first fashion comprises playing the audio at a first volume level, and wherein playing the audio in a second fashion comprises playing the audio at a second volume level that is different than the first volume level.
 10. A method of playing audio to a user and receiving input from the user when a voice application is being performed for the user, comprising: playing audio to the user; beginning to play one or more tones to the user in a first fashion once the audio has finished playing; detecting when the user begins to provide input after the one or more tones have begun playing; and playing the one or more tones without interruption, but in a second fashion that is different from the first fashion, when it is detected that the user has begun providing input and before the input is interpreted or checked for validity.
 11. The method of claim 10, wherein detecting when the user begins to provide input comprises detecting when a user begins providing spoken input.
 12. The method of claim 11, further comprising attempting to interpret the user's spoken input.
 13. The method of claim 12, wherein if the user's spoken input is successfully interpreted, the method further comprises stopping the playing of the one or more tones.
 14. The method of claim 12, wherein if the user's spoken input cannot be interpreted, the method further comprises again playing the one or more tones in the first fashion once the determination has been made that the user's spoken input cannot be interpreted.
 15. The method of claim 10, wherein playing the one or more tones in a first fashion comprises playing the one or more tones at a first volume level, and wherein playing the one or more tones in a second fashion comprises playing the one or more tones at a second volume level that is different than the first volume level.
 16. The method of claim 10, wherein the one or more tones comprise a repeating pattern of a plurality of tones, wherein playing the one or more tones in a first fashion comprises playing the repeating pattern of tones at a first tempo, and wherein playing the one or more tones in a second fashion comprises playing the repeating pattern of tones at a second tempo that is different from the first tempo.
 17. The method of claim 10, further comprising: detecting when a user begins to provide non-spoken input; and playing the one or more tones in a third fashion that is different from the first and second fashions when it is detected that the user has begun providing non-spoken input.
 18. The method of claim 17, wherein the non-spoken input is provided via a keyboard or a pointing device.
 19. A method of playing audio to a user when a voice application is being performed for the user, comprising: beginning to play a first type of audio to a user; detecting when the user begins to provide input; and continuing to play audio to the user without interruption, but playing a second type of audio to the user when it is detected that the user has begun providing input and before the input is interpreted or checked for validity.
 20. The method of claim 19, wherein detecting when the user begins to provide input comprises detecting when a user begins providing spoken input.
 21. The method of claim 20, further comprising attempting to interpret the user's spoken input.
 22. The method of claim 19, wherein detecting when the user begins to provide input comprises detecting when the user begins to provide non-spoken input. 